What phone do you have?
On Fri, 16 Jul 2004 11:59:39 +0500, atif <[EMAIL PROTECTED]> wrote: > I am configuring a sip-phone, receing calls, excellent voice quality. but it does > not place calls, please, can some one sort out. > > here is my debug output, and below that is sip-debug, > > Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to 0 > Jul 16 11:34:32 DEBUG[163850]: Stopping retransmission on 'iiasPlzFribMJMcW' > fesponse 1: Found > Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to 0 > Jul 16 11:34:32 DEBUG[163850]: Stopping retransmission on 'UCWmUU1tF0s6roEx' of > Response 2: Found > Jul 16 11:34:47 DEBUG[163850]: Auto destroying call 'iiasPlzFribMJMcW' > Jul 16 11:34:47 DEBUG[163850]: Auto destroying call 'UCWmUU1tF0s6roEx' > > ******SIP-DEBUG****** > Sip read: > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.187:5060;branch=z9hG4bKAtoCh7hGS > Max-Forwards: 70 > From: chinee <sip:[EMAIL PROTECTED]>;tag=Zlq179E4Jf8KX2lB > To: 13 <sip:[EMAIL PROTECTED]> > Call-ID: 1e020TNnX5IvcvFu > CSeq: 1 INVITE > Contact: <sip:[EMAIL PROTECTED]:5060> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 221 > > v=0 > o=- 0 0 IN IP4 192.168.0.187 > s=- > c=IN IP4 192.168.0.187 > t=0 0 > m=audio 1400 RTP/AVP 0 8 4 18 0 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:4 G723/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:0 telephone-event > > 12 headers, 11 lines > Using latest request as basis request > Sending to 192.168.0.187 : 5060 (non-NAT) > Found RTP audio format 0 > Found RTP audio format 8 > Found RTP audio format 4 > Found RTP audio format 18 > Found RTP audio format 0 > Peer RTP is at port 192.168.0.187:0 > Found description format PCMU > Found description format PCMA > Found description format G723 > Found description format G729 > Found description format telephone-event > Capabilities: us - 0x10f(G723|GSM|ULAW|ALAW|G729A), peer - > audio=0x109(G723|ALAW|G729A)/video=0x0(EMPTY), combined - 0x109(G723|ALAW|G729A) > Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) > Reliably Transmitting (no NAT): > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 192.168.0.187:5060;branch=z9hG4bKAtoCh7hGS > From: chinee <sip:[EMAIL PROTECTED]>;tag=Zlq179E4Jf8KX2lB > To: 13 <sip:[EMAIL PROTECTED]>;tag=as51de164a > Call-ID: 1e020TNnX5IvcvFu > CSeq: 1 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > Proxy-Authenticate: Digest realm="asterisk", nonce="50b81cdd" > Content-Length: 0 > > to 192.168.0.187:5060 > Scheduling destruction of call '1e020TNnX5IvcvFu' in 15000 ms > Found user 'chinee' > > Atif > > ________________________________________________________________ > Sent via the WebMail system at convergence.com.pk > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
